Method of providing voice over IP at predefined QoS levels

ABSTRACT

The present invention is a technique for provisioning and assessment of voice quality in Internet Protocol (IP) voice applications. A predetermined quality of service (QoS) is received from a user in the form of an acceptable packet loss. The technique measures current packet loss and delay in the network and in the terminal, and determines an optimum play-out delay for use by a play-out adaptation routine. An actual QoS is reported back to the user. To enhance packet delay and loss measurement, a QoS engine component is placed on the terminal side of the packetizer, therefore including packetizer losses in any measurement of the network.

FIELD OF THE INVENTION

The present invention relates generally to providing telephony and othercommunications services over a packetized network such as the Internet,and more specifically to a non-invasive, real-time technique forassessing and providing voice quality in Internet Protocol (IP) voiceapplications.

BACKGROUND OF THE INVENTION

As the use of the Internet increases, there is a concomitant increase inthe use of voice-over-IP (VoIP); i.e., the use of the Internet totransmit real-time voice conversations. This is attributed to theconvergence of computing and telecommunications under a single umbrella.Given availability of the requisite bandwidth, advanced users opt forthe immediacy of packet voice while engaged in an IP data application,as opposed to deriving voice associated with the application either fromanother medium, or at a later time.

The public Internet is a multi-node matrix of routers and switchesjoined by transport lines of differing capacities. As such, IP packetsmay experience processing delays at various nodes as they traverse theInternet matrix from one end-point to the next. Path differences canalso lead to variations in arrival times of the IP voice packets, andthis phenomenon, exaggerated by network congestions or other conditions,can have an adverse impact on the reconstruction of a voice conversationin real time. Link outages and traffic overload at specific nodes canalso lead to packet losses, with a greater potential for negativeimpacts on the service. Anomalies in the behavior of the integratednetwork have therefore had limiting effects on the quality of IP voiceapplications.

To address those problems, standards bodies have improved protocolspecifications, allowing greater predictability of the quality ofservice that a given application might support. However, beyond theseprotocol specifications, it is necessary to develop tools that areeffective in defining and gauging service quality. For example, it wouldbe desirable to provide a basis for raising customer acceptance levels,thus leading to the levels of confidence that are required for massdeployment.

VoIP is a real-time conversational application, and in any onedirection, the associated IP packet flow may be described as a real-timeisochronous media stream. To maintain quality and media coherency inisochronous applications, strict time dependencies between theapplication bits must be observed within the destination end-system. Theobjective metrics that are generally used to describe the packettransport characteristics between two end-points within the Internet aredelay, delay variation (also referred to as jitter), and packet loss.Delay refers to the time required to transmit an IP packet between twoend-points within the Internet. Causes of delay include processingoperations at routers within the network, increases in traffic load onthe network side, and coding and packetization processing on theterminal side. Jitter refers to variations in the packet inter-arrivaltimes from one end-system application to the next. Jitter is caused byfluctuations in network load, and differences in path routing ofindividual packets. Packet loss may refer either to arrivals that are solate as to render the packets unusable, or to the actual loss ofpackets. Packet loss is caused by network congestion, such asoverloading at routers.

It is important to be able to control those end-to-end transport metricsto achieve quality VoIP. The capability to assess the state of thosemetrics also directly leads to the capability to set and predict thequality of service (QoS) that is supportable between two end-pointswithin the Internet.

Much complexity is involved in quantifying the relationship between theraw end-to-end packet transport metrics and the perceived voice qualityfor a sample instantiation of IP voice. One viable approach to applyingthose transport metrics in determining VoIP QoS is a translation to MeanOpinion Scores (MOS) through the use of the E-Model. The E-Model isdescribed in detail in ITU-T Recommendation G.107, “The E-Model, aComputational Model for Use in Transmission Planning” (December 1998),the contents of which is incorporated by reference herein in itsentirety. MOS modeling has been in use for several years, and provides apsychological measure of voice quality. MOS scores are derived from thearithmetic average of a group of subjective responses. The technique iswidely adopted for voice quality assessment. The E-Model is an analyticmodel of voice quality for use in network planning purposes. The E-Modelprovides a method for estimating the relative voice quality whencomparing two reference connections. A highlight of the E-Model is thecomputation of the R-factor, which is used as a measure of voicequality. Once the R-factor is computed, the E-Model allows forstraightforward mapping back to MOS scores. The R-factor itself iscomputed by methods using the network transport metrics as discussed inCole, R. G. and Rosenbluth, J. H., “Voice over IP PerformanceMonitoring,” Computer Communication Review, V. 31, No. 2 at 9-14 (April2001), the contents of which is incorporated by reference herein in itsentirety.

The most common prior approach to prediction of QOS in the Internetinvolved the use of injecting echo probes into the network, and usingthe responses (or lack thereof) to those probes to measure the loss andround trip time (RTT). The RTT may then be used as a basis forestimating delay and jitter. The one-way delay is taken as one-half themeasured RTT. That technique does not support in-service VoIP assessmentand provisioning. Furthermore, the technique of injecting echo probesderives its metric information through the interface to routers in thenetwork, and is therefore not amenable to implementation within theend-system or to integration with a VoIP application.

It is therefore desirable to provide a method and system for evaluatinga quality of service (QoS) level for a communications service throughmeasurement of the network transport metrics available at theend-system, and for modifying parameters to achieve a QoS level based oncustomer requirements.

SUMMARY OF THE INVENTION

The present invention addresses those needs by providing a method forevaluating a quality of service (QoS) level for a communications serviceprovided over a packetized network, and modifying parameters of thatservice to achieve the QoS level.

One embodiment of the invention is a method for controlling acommunications terminal communicating over a packetized network. Amodule performing the method receives a quality of service requirementfor the service, and at least one transport metric describingcharacteristics of the network. Based on those parameters, the moduledetermines control information for use by a play-out adaptation routine.The control information is then transmitted to the play-out adaptationroutine.

In that method, the step of receiving a quality of service requirementfor the service may comprise receiving an attribute of acceptable packetloss. The step of receiving at least one transport metric describingcharacteristics of the network may comprise receiving at least one ofthe attributes of a group consisting of packet delay, jitter and packetloss.

The method may also include the step of receiving an error concealmentalgorithm. The algorithm may be based on a speech coder used in thecommunications terminal.

The step of determining control information may include estimating apacket loss distribution of the network. To estimate packet lossdistribution of the network, the module may measure actual packet loss.The estimation of packet loss distribution may also include adaptingparameters of a Pareto distribution.

The step of determining control information may include determining aplay-out delay for use by the play-out adaptation routine. The play-outdelay d may be defined:d=F ⁻¹(T)

wherein the function F⁻¹( ) is an inverse of a function defining aPareto distribution characterizing packet delays in the network, and Tis a target rate of packet accumulation required by the play-outadaptation routine to achieve the quality of service requirement. Thefunction defining a Pareto distribution characterizing packet delays inthe network may be:

${{F(x)} = {1 - \left( \frac{k}{x} \right)^{\alpha}}},{x \geq k}$

wherein estimates are used for the values of k and α:{circumflex over (k)}=min(x ₁ ,x ₂ ,x ₃ , . . . x _(n)) and

$\overset{\Cap}{\alpha} = {n\left\lbrack {\sum\limits_{i = 1}^{n}\;{\log\left( \frac{x_{i}}{\overset{\Cap}{k}} \right)}} \right\rbrack}^{- 1}$

where n is a total number of actual packet delay measurements and (x₁,x₂, x₃, . . . x_(n)) are actual packet delay measurements.

Further, the target rate T of packet accumulation required by theplay-out adaptation routine to achieve the quality of servicerequirement may be defined as:T=1−e _(playout)

wherein

${e_{playout} = \frac{e_{mos} - e_{network}}{1 - e_{network}}};$

e_(mos) being a packet loss probability associated with the quality ofservice requirement, and e_(network) being a packet loss probability ofthe network.

Another aspect of the invention is an audio terminal for use incommunication over a packetized network. The terminal includes apacketizer for extracting incoming audio data from packets received fromthe network, a play-out adapter for modulating a flow of audio data fromthe packetizer, and a quality of service engine between the play-outadapter and the packetizer. The quality of service engine is configuredto control the modulation of audio data by the play-out adapter based ontransport metrics of both the packetizer and the network.

The transport metrics may include at least one metric from the groupconsisting of average packet delay, packet delay variation and packetloss.

The audio terminal may further comprise a quality of service analysisroutine configured to receive the transport metrics from the quality ofservice engine, and to transmit play-out control information to thequality of service engine. The quality of service analysis routine mayfurther be configured to receive a quality of service requirement from anetwork API, and to transmit to a network API an achievable quality ofservice level. The play-out control information may include a play-outdelay.

The quality of service engine may control the modulation of audio databy the play-out adapter based on a packet loss probability e_(playout)of the play-out routine determined by

$e_{playout} = \frac{e_{mos} - e_{network}}{1 - e_{network}}$

wherein e_(mos) is a required maximum packet loss probability ande_(network) is the packet loss probability based on transport metrics ofboth the packetizer and the network.

The audio terminal may also include an audio controller for encoding anddecoding audio data exchanged with the play-out adapter; and a DTXengine for preparing audio data exchanged with the audio control fordiscontinuous transmission. The DTX engine interfaces with the qualityof service engine to coordinate the modulation of audio data.

These and other advantages of the invention will be apparent to those ofordinary skill in the art by reference to the following detaileddescription and the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a schematic view of an integrated network showing quality ofservice requirements distribution.

FIG. 2 is a schematic view of a communications terminal according to oneembodiment of the invention.

FIG. 3 shows a generalized protocol architecture for IP-basedcommunications.

FIG. 4 is a partial schematic view of parameter flow in a communicationsterminal according to the invention.

FIG. 5 is a block diagram depicting one method according to theinvention.

FIG. 6 is a table showing an exemplary mapping of R-factors to MOSvalues.

FIG. 7 is a table showing exemplary estimated encoding delays forvarious CODECs.

DETAILED DESCRIPTION

Referring to FIG. 1, a construct is shown in which quality-of-servicerequirements are distributed across the various segments of anintegrated network 100. A VoIP application 110 may require a specificend-to-end service quality 120 that might be resolved into threecomponents 121, 122, 125 that may be individually applied to the network135 and two end-systems 131, 132. Without the capability to control andassess the transport performance of the integrated network 100, thereare likely to be differences in the quality-of-service requested by theapplication, and that which is provided by the integrated network. Inthe method and apparatus of the invention, the routines for QoSdefinition and assessment are integrated within the end systems 131,132, and provide greater fidelity relative to the underlying end-to-endQoS 120 that is available to the VoIP application.

FIG. 2 is a diagram showing a generalized IP voice terminal 200 in whicha preferred embodiment of the QoS technique of the invention isincorporated. IP voice terminal software components 205 include apacketizer 210, a QoS engine 215, a play-out adaptation routine 220, anaudio control 225 and a DTX (discontinuous transmission) engine 230. TheIP voice terminal 200 interfaces with an IP network 250 through thepacketizer 210, which is an integral functionality of the terminal.Within the packetizer 210, in the direction toward the network, audioframes are encapsulated in packets for transmission using the real-timetransport protocol (RTP). In the direction away from the network 250,the packetizer 210 prepares packets received from the network forprocessing by the terminal.

The size of the audio packets sent to the network 250 has a directimpact on bandwidth utilization and the processing load within thepacketizer 210. Variations in the behavior of the packetizer cantherefore affect the measured QoS. In order to account for thatcomponent of QoS, the QoS engine 215 is placed between the play-outadapter 220 and the packetizer 210. Such a placement has the result ofcapturing the effects of packet delays within the network together withthe effects of the behaviour of the packetizer, offering a more precisemeasure of the end-to-end performance afforded by the overall system.

The QoS engine 215 measures transport metrics such as average delay,delay variation and packet loss. The QoS engine may work in conjunctionwith DTX.

The QoS engine 275 and DTX engine 230 are utilized through the use of anapplication programmer interface (API). FIG. 3 shows a generalizedprotocol architecture 300 for IP-based communications according to theinvention. As shown in FIG. 3, the network API interfaces with IP withreference to Network Time Protocol (NTP) 320, Real-Time TransportProtocol (RTP) 330 and the Real-Time Control Protocol (RTCP) 325. Thearchitecture is an integrated services architecture available to the IPvoice terminal for audio communications across the Internet. The RTP,RTCP and NTP protocols are utilized between the network API 310 and theInternet, through UDP/IP (Universal Datagram Protocol/Internet Protocol)350. NTP 320 specifies a procedure for time synchronization, relative tonational standard time within the Internet, through the use ofdistributed servers that operate in a self-organizing mode. RTP 330 is atransport protocol for real-time packet traffic. A relevant feature ofRTP is the inclusion of time stamps for use in media synchronization.RTCP 325 is used for media flow control.

FIG. 4 shows data exchanged among the various components of oneembodiment of the invention. A specific QoS requirement 415 may be inputinto the application through the network API 410 by a user. That input415 is directed to the QoS analysis routine 420. The QoS analysisroutine also takes as input the transport metrics 425 that are derivedfrom the computations in the QoS engine 450. Based on those data, theappropriate control information 430 is sent back to the QoS engine to bepassed on to the play-out adaptation routine. The QoS analysis routine420 also sends attainable MOS scores 440 back to the Network API fortransmittal to the calling routine/user.

In one example, shown in FIG. 2, an acceptable loss 271 is input to theQoS analysis routine 260 using a calling routine 270. The QoS analysisroutine 260 generates play-out delay information 265 using transportmetrics from the QoS Engine 215, and sends the play-out delayinformation to the play-out adaptation routine 220. The play-outadaptation routine uses the play-out delay data to adjust buffering. Theachievable MOS 272 is a return parameter to the calling routine 270. Inthe QoS analysis routine, the transport metrics received from the QoSengine may include the delay, jitter, and loss parameters. Thoseparameters are mapped to MOS scores through the use of the R-factormapping function. The MOS scores are then returned to a higher levelcalling routine 270.

The play-out adaptation routine 220 forwards buffered audio data to theaudio control module 225. The audio control module includes acoder/decoder (CODEC) 226 that contains specific algorithms forperforming compression and encryption according to the schemes in use bythe IP telephony system. As is known in the art, the terminal alsoincludes a discontinuous transmission (DTX) module that suspendstransmission during periods of silence, freeing bandwidth for otheruses.

A possible input into the QoS analysis routine 260 is an errorconcealment algorithm. The specific algorithm used depends on the typeof CODEC 226 used in the application. The error concealment scheme isdesigned to improve the performance of the application based on thepacket loss distribution within the network.

A method according to the invention is shown in FIG. 5. The method 500is preferably executed by a QoS analysis module that controls acommunications service over a packetized network. A quality of servicerequirement for the communications service is first received (step 505)by the terminal. As noted above, the QoS requirement may be an attributeof acceptable packet loss, such as a desired MOS input by a user. TheQoS requirement is preferably input into the QoS analysis module using acalling routine.

The module also receives (step 510) at least one transport metricdescribing characteristics of the network. The transport metric ismeasured using NTP, RTP and/or RTCP. It may be average delay, delayvariation, packet loss, or any combination of those metrics or any othermetrics that reflect characteristics affecting performance of thenetwork.

Once those values are received, the module determines (step 520) controlinformation for use by a play-out adaptation routine. That controlinformation is based on the quality of service requirement and on thetransport metric or metrics. In a preferred embodiment, that controlinformation includes a play-out delay. The determination of the controlinformation is described in more detail below.

The control information is subsequently transmitted (step 530) to theplay-out adaptation routine, either directly or through the QoS engine.

A key functionality of the QoS analysis routine 260 (FIG. 2) is theestimation of packet loss distribution for use both in definition andcontrol of the quality of service. In an illustrative embodiment, theloss distribution is computed by first computing running measurements ofpacket losses and delays within the network, using RTP in the terminal.The delay information is then used to estimate the parameters of ageneralized function that characterizes the delay distribution in thenetwork. In a preferred embodiment, the delay distribution isrepresented by a Pareto distribution. The parameters of the Paretodistribution are adapted to changes in traffic loads, so thattime-of-day variations may be incorporated. Network losses are recordedfor further use.

A specific QoS is realized by allocating loss during play-out. Theallocation depends on both the loss in the network and that which hasbeen scheduled in the play-out adaptation routine. The combined loss inthe network and in the play-out adaptation routine is referred tohereinafter as “overall loss.” The loss that is allowed in the play-outroutine is “residual loss,” beyond that which has occurred in thenetwork, and will be allocated based on the use of the Paretodistribution.

In accordance with the present invention, let a given MOS be specifiedby a user through the calling routine. Let the associated packet lossprobability be given by e_(mos). e_(mos) shall be the acceptable lossprobability for the particular application. In the QoS engine, sampledelays will be measured through the use of RTP, over some specifiedinterval I, whereby N such measurements are collected. A lossprobability for the network, e_(network), is computed in the QoS engine.The acceptable residual loss that must be allowed in the play-outroutine to support the specified MOS is given by r=e_(mos)−e_(network).Of course, if the loss in the network is greater than what isestablished by the QoS requirement, then the specified quality ofservice would not be supported. The acceptable loss probability,e_(playout), that may be allowed in the play-out routine is then givenby:

$e_{playout} = \frac{e_{mos} - e_{network}}{1 - e_{network}}$

The Pareto distribution is described by the following equation:

${{F(x)} = {1 - \left( \frac{k}{x} \right)^{\alpha}}},{x \geq k}$

where estimates of the parameters k and α are given by:{circumflex over (k)}=min(x ₁ ,x ₂ ,x ₃ , . . . x _(n)) and,

$\overset{\Cap}{\alpha} = {n\left\lbrack {\sum\limits_{i = 1}^{n}\;{\log\left( \frac{x_{i}}{\overset{\Cap}{k}} \right)}} \right\rbrack}^{- 1}$

In the play-out adaptation procedure, play-out delay is adaptively setto ensure that the residual loss for a given QoS is achieved. Theplay-out delay may be tied to the required MOS as follows: The targetrate of packet accumulation that is needed by the speech play-outprocess to achieve the required MOS, given the losses that have occurredin network, is given by:T=1−e _(playout)

The play-out delay, d, required to achieve this QoS is then computed as:d=F ⁻¹(T)

where F⁻¹( ) is the inverse of F( ) defined above.

Within the QoS analysis block, the network transport metrics computed inthe QoS engine are mapped to MOS scores for perceptual comparison withwell-known circuit-switched implementations. The mapping procedure isfacilitated with CODEC-specific information, and use is made of theE-model. When using the E-model, it is necessary to first compute anR-factor using the network transport metrics. The R-factor is thenmapped to MOS values through the use of the information represented inthe table of FIG. 6.

The general equation for the R-factor is defined by:R=100−I _(s) −I _(d) =I _(ef) +Awherein I_(s), I_(d), and I_(ef) are used to denote the impairmentsassociated with the signal-to-noise ratio, network delay, and packetlosses respectively. The parameter A is a normalization factor. For aspecific application, the R-factor may be reduced to:R=β−I _(d) −I _(ef)

wherein β, the reduction factor, may be fixed for pure packet-switchednetworks, or networks involving a combination of both packet-switchedand circuit-switched networks. In one embodiment of the presentinvention in which default parameters are assigned values consistentwith those described in ITU-T Recommendation G.107, β takes on a valueof 94.2.

An analytic expression for the delay impairment, I_(d), is given inITU-T Recommendation G.107. However, invoking the assumption of an allIP environment, and applying default parameters as provided by theRecommendation, that expression may illustratively be reduced to:I _(d)=0.024d+0.11(d−177.3)H(d−177.3)

wherein d is the one-way delay as defined by:d=d _(codec) +d _(playout) +d _(network)

and the function H(x) is defined as:

${H(x)} = \left\{ \begin{matrix}{0,} & {{{{for}\mspace{14mu} x} < 0},} \\{1,} & {{{for}\mspace{14mu} x} > 0}\end{matrix} \right.$

d_(codec) in the above definition of d is a processing delay associatedwith the specific speech coder in use in the application. That delaycharacterizes the encoding, compression, and packetization processes.Examples of the CODEC-specific delays for two common CODECs areillustrated in FIG. 7, wherein N is the number of 10 millisecond framesthat are packetized in a single IP frame.

The expression for I_(ef), the impairment associated with packet loss,is normally determined from MOS characterization of the available CODECsunder various operating conditions. Given a number of sample datapoints, a curve fitting procedure is conducted to determine theparameters, γ₁ . . . γ_(i), of the following curve:I _(ef)=γ₁+γ₂ ln(1+γ₃ e)

where e is the total loss probability, and is given by:e=e _(network)+(1−e _(network))e _(playout)

The fitting parameters, γ_(i), are CODEC-specific. As an illustration,for a G.729a CODEC with a packet size of 20 ms, and random packet lossof up to 16%, I_(ef) is given by:I _(ef)≈11+40 ln(1+10e)

Using the reduced definition of the R-factor presented herein allows thedevelopment of a numerical expression for R that can be mapped to an MOSvalue by the application of the information presented in the table ofFIG. 6.

In accordance with the present invention, the proposed technique can beimplemented in a number of areas. For example, the technique of theinvention may be used for service characterization, whereby a terminalhaving the described functionality is deployed in a specific locationand used to communicate with similar terminals to characterize the VoIPQoS. In another example, the inventive technique may be used for faultmanagement in situations where network problems exist, and the sourcesof those problems are to be localized.

Further, the method of the invention may be used for base-linemonitoring in which, after enhanced QoS policies are applied, thetechnique is used to ensure that such enhancements are available. Thetechnique may also be used for performance monitoring in cases, forexample, where a service level agreement (SLA) is in place.

The foregoing Detailed Description is to be understood as being in everyrespect illustrative and exemplary, but not restrictive, and the scopeof the invention disclosed herein is not to be determined from theDetailed Description, but rather from the claims as interpretedaccording to the full breadth permitted by the patent laws. It is to beunderstood that the embodiments shown and described herein are onlyillustrative of the principles of the present invention and that variousmodifications may be implemented by those skilled in the art withoutdeparting from the scope and spirit of the invention.

1. A method for controlling a communications terminal communicatingaudio data over a packetized network, comprising the steps of: receivinga quality of service requirement for a service, wherein the step ofreceiving the quality of service requirement for the service comprisesreceiving an attribute of acceptable packet loss; receiving at least onetransport metric describing characteristics of the network; determiningcontrol information for use by a play-out adaptation routine, thecontrol information being based on the quality of service requirementand on the at least one transport metric; and transmitting the controlinformation to the play-out adaptation routine to adapt the audio datain accordance with the control information.
 2. A method for controllinga communications terminal communicating audio data over a packetizednetwork, comprising the steps of: receiving a quality of servicerequirement for a service; receiving at least one transport metricdescribing characteristics of the network; determining controlinformation for use by a play-out adaptation routine, the controlinformation being based on the quality of service requirement and on theat least one transport metric; transmitting the control information tothe play-out adaptation routine to adapt the audio data in accordancewith the control information; and receiving an error concealmentalgorithm.
 3. The method of claim 2, wherein the error concealmentalgorithm is based on a speech coder used in the communicationsterminal.
 4. A method for controlling a communications terminalcommunicating audio data over a packetized network, comprising the stepsof: receiving a quality of service requirement for a service; receivingat least one transport metric describing characteristics of the network;determining control information for use by a play-out adaptationroutine, the control information being based on the quality of servicerequirement and on the at least one transport metric, wherein the stepof determining control information includes estimating a packet lossdistribution of the network; and transmitting the control information tothe play-out adaptation routine to adapt the audio data in accordancewith the control information.
 5. The method of claim 4, wherein theestimation of packet loss distribution of the network includes measuringactual packet loss.
 6. The method of claim 4, wherein the estimation ofpacket loss distribution of the network includes adapting parameters ofa Pareto distribution.
 7. A method for controlling a communicationsterminal communicating audio data over a packetized network, comprisingthe steps of: receiving a quality of service requirement for a service;receiving at least one transport metric describing characteristics ofthe network; determining control information for use by a play-outadaptation routine, the control information being based on the qualityof service requirement and on the at least one transport metric, whereinthe step of determining control information includes determining aplay-out delay d for use by the play-out adaptation routine, wherein theplay-out delay d is defined:d=F ⁻¹(T) wherein the function F⁻¹( ) is an inverse of a functiondefining a Pareto distribution characterizing packet delays in thenetwork, and T is a target rate of packet accumulation required by theplay-out adaptation routine to achieve the quality of servicerequirement; and transmitting the control information to the play-outadaptation routine to adapt the audio data in accordance with thecontrol information.
 8. The method of claim 7, wherein the functiondefining the Pareto distribution characterizing packet delays in thenetwork is:${{F(x)} = {1 - \left( \frac{k}{x} \right)^{\alpha}}},{x \geq k}$wherein estimates are used for the values of k and α:{circumflex over (k)}=min(x ₁ ,x ₂ ,x ₃ , . . . x _(n)) and$\overset{\Cap}{\alpha} = {n\left\lbrack {\sum\limits_{i = 1}^{n}\;{\log\left( \frac{x_{i}}{\overset{\Cap}{k}} \right)}} \right\rbrack}^{- 1}$where n a total number of actual packet delay measurements and (x₁, x₂,x₃, . . . x_(n)) are actual packet delay measurements.
 9. The method ofclaim 7, wherein the target rate T of a packet accumulation required bythe play-out adaptation routine to achieve the quality of servicerequirement is defined as:T=1−e _(playout) wherein${e_{playout} = \frac{e_{mos} - e_{network}}{1 - e_{network}}};$ e_(mos)being a packet loss probability associated with the quality of servicerequirement, and e_(network) being a packet loss probability of thenetwork.
 10. An audio terminal for use in communication over apacketized network, the terminal comprising: a packetizer for extractingincoming audio data from packets received from the network; a play-outadapter for modulating a flow of audio data from the packetizer; aquality of service engine between the play-out adapter and thepacketizer, the quality of service engine being configured to controlthe modulation of audio data by the play-out adapter based on transportmetrics of both the packetizer and the network; and a quality of serviceanalysis routine configured to receive the transport metrics from thequality of service engine, and to transmit play-out control informationto the quality of service engine, wherein the quality of serviceanalysis routine is further configured to receive a quality of servicerequirement from a network API, and to transmit to the network API anachievable quality of service level.
 11. An audio terminal for use incommunication over a packetized network, the terminal comprising: apacketizer for extracting incoming audio data from packets received fromthe network; a play-out adapter for modulating a flow of audio data fromthe packetizer; and a quality of service engine between the play-outadapter and the packetizer, the quality of service engine beingconfigured to control the modulation of audio data by the play-outadapter based on transport metrics of both the packetizer and thenetwork, wherein the quality of service engine controls the modulationof audio data by the play-out adapter based on a packet loss probabilitye_(playout) of a play-out routine determined by${e_{playout} = \frac{e_{mos} - e_{network}}{1 - e_{network}}};$ whereine_(mos) is a required maximum packet loss probability and e_(network) isthe packet loss probability based on transport metrics of both thepacketizer and the network.